Communication services today are generally based on either the public switched telephone network (PSTN), which provides circuit-switched voice servers, real-time faxing, and central office (CO)-based voice mail, or the Internet protocol (IP)-based Internet, intranet, or extranet systems, which provide services such as e-mail.
The PSTN and the IP-based systems rely on different architectures and protocols. The PSTN generally refers to a collection of interconnected systems operated by the various telephone companies and post telephone and telegraph administrations (PTTs) around the world. PSTN systems use dedicated circuits. That is, when a phone number is dialed, switches in the network lock-up a dedicated circuit that can only be used by that phone call and exist until that phone call is terminated. Thus, even when there is no information to carry as part of the transmission (e.g., the callers are silent) the circuit is still in use by the call.
Instead of sending transmissions over a dedicated circuit, IP-based systems packetize the transmissions with self-contained addresses that are then uploaded through a gateway into the network where they are relayed between routers until they reach their final destination. This packetization of transmissions allows for much more efficient transport of transmissions since empty space in a transmission (where no information is present) can be filled by additional transmissions.
Traditional telecommunications carriers and service providers are moving from PSTN to packet-based IP infrastructure. The contents offered over IP include, for example, voice, fax, instant messaging, digital television, etc. In the case of voice, the technology is referred to as voice over IP (VOIP).
One aspect for a telephony network may be call control, including call setup and teardown. As VOIP operates in the absence of dedicated circuits, call control may be used to avoid service outage, quality of service (QoS) degradation, and loss of revenue. Call control may be achieved via Session Initiation Protocol (SIP).
SIP is a signaling protocol using text messages. There are two basic SIP entities: SIP user agents (UAs) and SIP servers. SIP servers may further be grouped into proxy servers for session routing and registration servers for UA registration.
SIP is an application level protocol that may run on top of a transport layer, such as the transmission control protocol (TCP) or user diagram protocol (UDP).
SIP is widely used for setting up and tearing down multimedia communication sessions such as voice and video calls over the Internet. Other SIP application examples include video conferencing, streaming multimedia distribution, instant messaging, presence information and online games. For example, using SIP for instant messaging and presence implies an additional load on the SIP based call control infrastructure.
In November 2000, SIP was accepted as a 3rd Generation Partnership Project (3GPP) signaling protocol and permanent element of the IP multimedia subsystem (IMS) architecture for IP based streaming multimedia services in cellular systems. This development means that the control infrastructure of the SIP-based call control infrastructure needs to handle terminal mobility between different cell boundaries, in addition to call setup and teardown. Both the support for mobile users and hand-off represents additional loads on the call control infra-structure.